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SIP Server now offers the option to use SIP Feature Server as an “external dial plan” as an alternative to the internal SIP Server dial plan. Each choice offers distinct advantages to consider when choosing which dial plan to use. (Note that dial plans may not be combined.)

Flexible rules with pattern matching logic for choosing a trunk for outgoing calls

Enhanced support for deployments where voicemail mailboxes are assigned to users (but not to DNs)

SIP Server Dial Plan Highlights:

Many supported parameters for advanced dial-plan rules, such as “onbusy”, “type”, “calltype”, “clir”, and more

Native support by SIP Server (smaller footprint, less complexity if Feature Server is not required for the deployment)

SIP Server offers additional control over how a dial plan is applied to the destination of TRouteCall and/or to multi-site (ISCC) calls that are routed through an External Routing Point with two configuration options:

The rp-use-dial-plan configuration option changes the default behavior of the dial plan to any one of the following:

The rp-use-dial-plan option applies to both SIP Server and SIP Feature Server dial plans. If the UseDialPlan key-value pair is present in AttributeExtensions of TRouteCall, then it takes priority over the rp-use-dial-plan option.

The enable-iscc-dial-plan option enables SIP Server to apply the dial plan to the target destination when a call is routed from an External Routing Point to a DN at the destination site.

Under a SIP Server Switch object that is associated with the SIP Server, create a VOIP Service DN named fs-dialplan and configure these options:

service-type—Set this option to extended.

Important: Ensure that you add the final slash character (/) to the end of each of the following URLs.

url—Set this option to http://FS Node:port/

For n+1 High Availability (HA), add the following parameters:

url-1 = http://FS Node2:port/

url-2 = http://FS Node3:port/

url-n = http://FS Node_N:port/

Important: A Feature Server's dial plan URL must be configured only on a VOIP Service DN that was created on the Switch controlled by the SIP Server that is connected to that particular Feature Server.

If required, configure the following options in the SIP Server Application object, the [TServer] section:

rp-use-dial-plan

default—For a SIP Server dial plan, the same as the false value. For a Feature Server dial plan, the same as the partial value.

full—The dial plan is applied to the destination of TRouteCall, including the digit translation and forwarding rules.

partial—Only the digit translation is applied to a dial-plan target. Forwarding rules, such as forwarding on no answer (ontimeout), forwarding on busy (onbusy), forwarding on DND (ondnd), forwarding on no response (onunreach), and forwarding on not SIP registered (onnotreg) are not applied. Valid for both SIP Server and SIP Feature Server dial plans.

false—No dial plan is applied to the destination of TRouteCall.

Important

If the SIP Server dial plan is used, SIP Server selects the dial plan assigned to the caller. This is the dial plan configured for the DN/Agent Login of the DN for internal calls, or the Trunk DN for inbound calls, or the Application-level option if no DN/Agent-Login-level dial plan is configured.

enable-iscc-dial-plan

Specifies whether SIP Server applies the dial plan to the agent destination of multi-site (ISCC) calls that are routed through an External Routing Point (cast-type=route-notoken), as follows:

If set to true, the dial plan (full, including the digit translation and forwarding rules) is applied.

If set to false, the dial plan is not applied.

This option must be configured on the remote (destination) site. SIP Server applies the dial plan when a call is routed from an External Routing Point to a DN at the destination site.

Important

SIP Server will still apply the dial plan to the External Routing Point destination of multi-site (ISCC) calls, and this will take priority over the agent DN destination dial-plan rule regardless of the setting of enable-iscc-dial-plan.

AttributeExtensions

full—The dial plan is applied to the destination of TRouteCall, including the digit translation and forwarding rules.

partial—Valid for both SIP Server and SIP Feature Server dial plans. Only the digit translation is applied to a dial-plan target. Forwarding rules, such as forwarding on no answer (ontimeout), forwarding on busy (onbusy), forwarding on DND (ondnd), forwarding on no response (onunreach), and forwarding on not SIP registered (onnotreg) are not applied.

false—No dial plan is applied to the destination of TRouteCall.

Important

For ISCC calls, this extension is applied only to calls routed through an External Routing Point (cast-type=route-notoken).